Call routing and real-time monitoring

ABSTRACT

A method and system for providing call routing analytics. A virtual session initiation protocol switch is provided and hosted in an Internet cloud-based environment. The switch streams live call detail records to a computer system having a processor configured to process all of the subscriber&#39;s call records to monitor route performance for the subscriber. Real-time route performance data is transmitted to the subscriber for display at a subscriber computer. The subscriber can then alter a routing of at least a portion of the call utilizing the switch in response to the real-time route performance data to increase quality of signaling and business performance.

CROSS REFERENCE TO RELATED APPLICATION

This application claims the benefit of provisional patent applicationSer. No. 61/766,845, filed on Feb. 20, 2013, the entire contents ofwhich are incorporated herein by reference.

FIELD

The present invention relates generally to telecommunications, andparticularly to a system and method for providing telephone call routinganalytics for a telecommunications carrier.

SUMMARY

The present invention is directed to a method for providing call routinganalytics comprising providing a virtual session initiation protocolswitch that is hosted in an internet cloud-based environment, providinga processor programmed to extract call detail records for a subscriberfor all calls that utilize the switch, automatically processing all ofthe subscriber's call records to monitor route performance for asubscriber, transmitting real-time route performance data to thesubscriber for display at a subscriber computer, and altering a routingof at least a portion of the calls utilizing the switch in response tothe real-time route performance data.

The invention is further directed to a system for providing call routinganalytics, comprising a cloud hosted session initiated protocol switch,wherein the switch streams call detail records, a call detail recordqueue to temporarily hold the call detail records, a processorprogrammed to divide the call detail records into analytic constituents,transmit call detail records to a storage media, process the analyticconstituents and transmitting real-time route performance data to asubscriber for display at a subscriber computer based on the analyticconstituents; and a subscriber interface configured to receive theanalytic constituents.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a diagrammatic representation of a system for providing callrouting analytics to a subscriber.

FIG. 2 is a diagrammatic representation of an Internet web page showingcall routing analytics provided to a subscriber at a subscriberinterface.

FIG. 3 is an illustration of call routing analytics provided asubscriber showing detailed routing information on a per carrier basis.

FIG. 4 is an illustration of an Internet web page that utilizes thesystem and method of the present invention to provide a subscriber withcertain call routing analytics.

DESCRIPTION

In telecommunications networks, Session Initiation Protocol (SIP) is anapplication-laver signaling protocol used to create, modify, andterminate telephone calls haying one or more users. These telephonecalls include cell phone to cell phone, land line to land line, landline to cell phone and Internet telephone calls. SIP has become a widelyaccepted IP signaling service in the telecommunications industry.

The main function of an SIP switch is to initiate and terminateinteractive communications between users. The SIP switch performsseveral functions such as determining the location of users, monitoringcall length, call attempts and completions, call minutes, the averagesuccess rate of calls from one carrier to another, and post dial delay.In operation a telephone information signal is transmitted from anorigin device, telephone, through the origin user's carrier to the SIPswitch. The SIP switch passes the telephone information signal to thedestination user's carrier. When the destination user is located the SIPswitch transmits response to the call invitation to the originatinguser. If the invitation is accepted the call is connected and becomesactive. When the call is concluded, the SIP switch terminates the callconnection. A call from an IP enabled device to a standard telephone setconnected to the Public Switched Telephone Network (PSTN) is one type ofcall that can be placed through the SIP switch. The IP enabled devicecommunicates with the PSTN via an IP network gateway. The networkgateway is an interface that provides mediation from a packet switchedIP signaling/voice system to the circuit switched, time divisionmultiplexed (TDM) system employed by the PSTN.

The accurate and timely billing of calls passed across the SIP switchand determining the routing performance of calls are important factorsto carriers for profitability and quality assurance. Each call thattransverses the SIP switch includes call metadata that may be collectedfrom the can to generate a call detail record (CDR). Call data recordsinclude information such as an SIP code, an SIP reason, an origin callerID, time stamp, call start, answer, and end time, call duration, postdial delay time, the origin carrier name, an origin. trunk name, theorigin billing rate, and origin prefix. The metadata may also includethe terminating carrier, the terminating trunk name, the terminatingcarrier's billing rate and time billed, the jurisdiction of the call,the switch IP address, and several other characteristics of the call.This data is used to generate billing records for carriers to use incollecting fees for use of their services. This metadata can alsoprovide the carrier with call routing analytics that are useful indetermining the efficiency of the carrier's network and the efficiencyand cost of networks the carrier is utilizing to connect calls.

In addition to telephone calls, SIP sessions may be used in videoconferencing, streaming multimedia distribution, instant messaging,presence information, file transfer, fax over IP, online games, etc. TheSIP protocol governs the establishment, termination and other elementsof a call made over IP networks. Thus, SIP switches may create abottleneck for messages sent between carriers that are used to connectcalls.

Voice-over-IP (VoIP) carrier connectivity includes numerous sessionborder controllers or SIP proxies that act as a gateway on the edge of acarrier network and mediate incoming and outgoing traffic from othercarriers. Voice carriers are responsible for interconnecting phonenetworks worldwide so that phone networks on different platforms andwith competing business frameworks can complete calls in a way that isseamless to the end-user. The SIP switch of the present inventionprovides a telecommunications exchange that mediates transactions androutes calls seamlessly to a less expensive and more efficient carrier.The SIP switch may setup and complete the calls anonymously between thecarriers. Subscribers are able to conduct voice traffic termination overa distributed and shared architecture, without having any knowledge ofany other subscriber to the system. To the subscriber, nothing isshared. To the operator of the fabric, everything is shared.

The key components of an SIP proxy are voice switching, routing(including jurisdictional determination), CDR generation and storage,and security. The switching features may relate to the ability ofdifferent pieces of hardware to exchange voice signaling cooperativelyand to control the flow of traffic between carrier networks. The routingfeatures may relate to how calls are handled once they are inside theswitch and to which carriers a particular phone number is routed. TheCDR serves as both a method to bill a telephone call, but also containdiagnostic and analytic related information that may be used to monitorthe caller's performance. The security function of the switch may act asa gatekeeper and may permit only carriers who have specificallyinterconnected to pass traffic.

A traditional SIP proxy or switch handles the above described functionswith a single piece of hardware (SIP Server) deployed in a fixedlocation with an independently maintained device. Because, a traditionalSIP switch is a hardware component, it may become a bottleneck toefficient and effective operation of the network when network trafficexceeds the SIP hardware capacity.

The present invention provides a virtual SIP switch that segregates eachof the routing components and the software infrastructure allowing forhorizontal growth of switching capacity to alleviate the bottleneckdiscussed above. The switching, security, CDR storage and routingcomponents of the cloud-based SIP switch of the present invention areeach handled by clusters of purpose driven hardware and software suitesallowing deployment of a scalable voice routing fabric spread overmultiple geographic areas and shared by groups of subscribers. The SIPswitch of the present invention provides for the scalability of virtualvoice infrastructure and makes the voice fabric elastic. Theinfrastructure is self-healing and auto-expandable, where any clustercan sense the failure of a particular node and shift the resources tofill in the gap. Further, any cluster can expand the pool of availableresources based upon a cluster demand and provide additional cloudinfrastructure to handle the increased load. The clusters can begeographically diverse and location aware, where the origination ortermination IP address endpoints of a call may be used to geographicallychoose the closest cluster to either end of the call. This lowers theoverall latency, chance of packet loss, and round trip hops in a VoIPsignaled call.

Turning now to the figures, FIG. 1 shows a diagrammatic representationof a system 10 for providing call routing analytics to a subscriber. Thesystem comprises the cloud-based SIP switch 12 discussed above, a CDRqueue 14, a processor 16, and a subscriber interface 18. The CDR queuecomprises a read/write digital medium that receives CDR's streamed fromthe switch 12, temporarily stores the CDRs before passing them to theprocessor 16. The switch 12 streams a CDR for each call that crosses theswitch. Thus, thousands of CDRs may be streaming to the queue persecond. Because the queue 14 is cloud-based it may be expanded orcontracted to suit the amount of CDR data streaming from the switch.

The system uses processor 16 to divide the call detail records intoanalytic constituents and transmits them to the analytics queue 20. Theprocessor also copies each of the call detail records and routes themdirectly to a persistent database 22 for storage in accordance withapplicable industry and regulatory standards. Separate modules 24, 26,and 28 process the analytic constituents and transmit real-time routeperformance data to a subscriber for display at a subscriber interface.The modules may be configured to determine profitability of thesubscriber's traffic, number of calls, success rate of calls, vendormakeup, origin and destination of calls.

As used herein analytic components may comprise call characteristicsfrom the detail records that are used to determine a carrier's routeperformance. The processor may access data loaded into the system by asubscriber, such as call rate tables, to assist in determining routeperformance. As used herein, route performance may comprise quality ofsignaling and business performance. Real-time routing performance datamay comprise real-time subscriber profitability and margin.

To determine subscriber profitability and margin the processor accessesthe subscriber's preloaded rate tables and overhead allocation data. Theprocessor determines the length of each call accessing the subscriber'snetwork from the call detail record, the applicable rate and determinesthe amount it will bill for the call. The subscriber's overhead mayinclude the cost of any origin or destination charges incurred inhandling the call. Using this information the processor determines theprofitability of each call. The processor determines the profitabilityof each call as it passes through the SIP switch and updates theprofitability calculation provided to the subscriber in real-time. Thus,the subscriber has an interface, for example a web interface, whichallows it to track route performance as calls are connected,disconnected or fail as they occur. As used herein real-time routeperformance data may comprise metrics such as length of calls, averagelength of calls, origin rates, destination rates, jurisdiction data,answer seizure ratio, origin, destination, and post dial delay. Thereal-time route performance data may also include successful andunsuccessful connections through a carrier and volume of calls to andfrom a carrier and or subscriber. With this information the subscribermay alter its performance parameters to reroute call traffic to a moreefficient or effective route, depending on the subscriber's goal. Thisimmediate access to real-time analytic data allows the subscriber tomaintain profitability while monitoring the quality of signaling throughthe SIP switch. The virtual nature of the system of the currentinvention also permits for the reallocation of resources to immediatelymeet the growing or contracting needs of the subscriber by the switchprovider without the need for additional hardware or capital expense bythe subscriber.

The information is presented to the subscriber at the user interface ina manner that permits the subscriber to react to its route performanceas it is occurring instead of using stale data from call detail recordsthat are minutes or hours old.

Turning now to FIG. 2, there is shown a representative webpage 30 todisplay the real-time rate performance of the subscriber's calls. Thewebpage comprises a plurality of features designed to assist subscribersin monitoring route performance. For example, the page of FIG. 2 showsthe subscriber the call minutes 32 it has sold for a particular dayversus a historical day, current and historical call attempts 34, thetotal minutes 36 used for the applicable day, the subscriber's real-timedaily margin as dollars 38 and a percentage 40. The may also provide thesubscriber with other data such as calls per second. The page alsoprovides the subscriber the ability to see detailed informationsupporting the information displayed on the dashboard 41 such ascarriers 42, routing 44, sales 46, and accounting 48 in real-time simplyby clicking on the applicable button. The page also provides thesubscriber a real-time gauge 50 showing the number of ports activelyused by the subscriber. This feature is important to provide thesubscriber the ability to monitor and see when the subscriber's trafficis reaching its maximum allocated capacity and proactively seekallocation of additional resources through the switch instead ofreacting to call failure and poor network performance after the fact.

Turning now to FIG. 3, there is shown therein an exemplary web page atthe user interface used to communicate information for a dialer and itsinteraction with five (5) carriers. The number of carriers could beexpanded to include an infinite number of carriers. However, for thepurpose of simplicity only five (5) carriers are shown. The analytic isinformation provided the subscriber may include call attempts, completedcalls, minutes used, answer success rate (ASR), average length of call(ALOC), post dial delay (PDD), revenue, cost, margin, and marginpercentage. This information is culled real-time from the streamed CDRsand constantly updated and provided to the subscriber interface. Thus,the subscriber is able to monitor the real-time route performance andadjust the routing of calls to more efficient, effective, or lessexpensive carriers. The present invention may be used to provide a leastcost routing system without requiring the subscriber to use stale dataand invest in expensive or unnecessary hardware.

With reference now to FIG. 4, there is shown a webpage showing trafficfor a carrier based on country. The page shows total call traffic andcall traffic by country. Again, this information is live real-time dataculled from the CDRs streamed from the SIP switch. The data provided thesubscriber is modifiable by the subscriber, but may include callattempts, complete calls, minutes sold, ASR, ALOC, PDD, revenue, cost,margin and margin percentage.

In operation the present invention provides a method for providingrouting analytics in an SIP telecommunications network. A virtualsession initiation protocol switch is provided. The switch is hosted inan Internet cloud-based environment and used to connect multiplecarriers in a telecommunications environment. A processor programmed toextract call detail records for a subscriber for all calls that utilizethe switch is provided and automatically processes all for thesubscriber's call records to monitor route performance for a subscriber.The real-time route perfomance data is transmitted instantly to thesubscriber from the processor for display at a subscriber computer. Thesubscriber then is able to alter the routing of at least a portion ofthe calls utilizing the switch in response to the real-time routeperformance data. A complete version of each call detail record isstored in a database for later access, if needed. The real-time routeperformance data may comprise, but is not limited to, the subscriber'squality of signaling or real-time profitability. Thus, the subscribermay alter the subscriber's rate table or routing of at least a portionof the calls to increase the subscriber's profitability. In accordancewith the method of the invention, a predetermined set of analyticoutputs may be selected. All of the subscriber's call records may beprocessed by the computer system having a processor to monitor routeperformance to automatically dividing the call detail records into aplurality of component data sets and processing the component data setsto generate a plurality of analytic outputs corresponding to thepredetermined set of analytic outputs indicative of route performance.

Although the present invention has been described with respect topreferred. embodiment, various changes and modifications may besuggested to one skilled in the art, and it is intended that the presentinvention encompass such changes and modifications as fall within thescope of this disclosure.

What is claimed is:
 1. A method for providing call routing analytics,comprising: providing a virtual session initiation protocol switch thatis hosted in an internet cloud-based environment; providing a processorprogrammed to extract call detail records for a subscriber for all callsthat utilize the switch; automatically processing all of thesubscriber's call records to monitor route performance for a subscriber;transmitting real-time route performance data to the subscriber fordisplay at a subscriber computer; and altering a routing of at least aportion of the calls utilizing the switch in response to the real-timeroute performance data.
 2. The method of claim 1 further comprisingarchiving a complete version of each call detail record on a database.3. The method of claim 1 wherein the real-time route performance datacomprises quality of signaling.
 4. The method of claim 1 wherein thereal-time route performance data comprises the subscriber's real-timeprofitability.
 5. The method of claim 4 further comprising altering therouting of at least a portion of the calls to increase the subscriber'sprofitability.
 6. The method of claim 1 further comprising altering acall rate table.
 7. The method of claim 1 further comprising selecting apredetermined set of analytic outputs; wherein automatically processingall of the subscriber's call records to monitor route performancecomprises automatically dividing the call detail records into aplurality of component data sets and processing the component data setsto generate a plurality of analytic outputs corresponding to thepredetermined set of analytic outputs indicative of route performance.8. A system for providing call routing analytics, comprising: a cloudhosted session initiated protocol switch; wherein the switch streamscall detail records; a call detail record queue to temporarily hold thecall detail records; a processor programmed to divide the call detailrecords into analytic constituents, transmit call detail records to astorage media, process the analytic constituents and transmittingreal-time route performance data to a subscriber for display at asubscriber computer based on the analytic constituents; and a subscriberinterface configured to receive the analytic constituents,
 9. The systemof claim 8 wherein the real-time routing performance data comprisessubscriber profitability and call traffic in real-time.
 10. The systemof claim 8 wherein the real-time routing performance data compriseslength of calls, origin rates, destination rates, jurisdiction data,answer seizure ratio, origin, destination, and post dial delay.
 11. Thesystem of claim 8 wherein the route performance comprises quality ofsignaling.
 12. The system of claim 8 wherein the subscriber interfacecomprises a secure website to display the real-time rate performance ofthe subscriber's calls.
 13. The system of claim 8 wherein the routeperformance comprises a business performance parameter of thesubscriber.